I am moving from a PRI connection to SIP/SIP, using a cube. In the PRI world, for inbound calls, I had a translation pattern on the imcoming pots dial peer, and simply used the "incoming called number ." to match all incoming calls.
when I configured the respective inbound DP's (one inbound from ISP, the other Inbound from Phones) SIP voip dial peers, I have
dial-peer voice 100 voip description Inbound from SIP translation-profile incoming 10_to_4_IN session protocol sipv2 incoming called-number . dtmf-relay rtp-nte ! dial-peer voice 2005 voip description Outbound from CUCM voice-class codec 2 no voice-class sip early-offer forced session protocol sipv2 incoming called-number . dtmf-relay rtp-nte sip-kpml sip-notify
How do I know or guarantee that inbound calls from the ISP will use the correct DP if both have "incoming called number . "? Also, is there an easier way to see DP matching other than debug voice dialpeer?
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...